Note: By default, G.729a is not enabled, and G711 codec calls are negotiated first. For more information about assigning codec preferences, see Details of Technician Commands information about the setipcodec command.

The protocol that is responsible for communication between Cisco Unified Communications Manager and the Cisco Unified MeetingPlace H.323/SIP IP Gateway. The protocol suite, which extends H.225 for call signaling and H.245 for data transfer, is used in the successful acceptance and media exchange of data.

Session Initiation Protocol (SIP)

A call-control protocol that supports all existing functionality that is available to a Cisco IP phone. Cisco Unified MeetingPlace H.323/SIP IP Gateway Release 5.3 complies with RFC 3261 and RFC 3515 specifications and interoperates with the following endpoints:

An Internet protocol responsible for the transmission of real-time data, such as video and audio. Generally, RTP runs on top of User Datagram Protocol (UDP) but can also be supported by other transport protocols.

A protocol that is used to establish connections, locate resources, forward data, and handle flow control and error recovery, which enable a Cisco IP phone to notify Cisco Unified Communications Manager of its ability to place and receive calls.

Cisco Unified MeetingPlace Gateway System Integrity Manager (SIM)

A messaging service that enables NT services on the IP-gateway server to communicate directly with the Cisco Unified MeetingPlace system.

Dual Tone Multi-Frequency (DTMF) is a signaling method that allocates a specific pair of frequencies to each key on a touch-tone telephone. Various Cisco Unified MeetingPlace Audio Server system functions are invoked when callers press touch-tone keys in certain combinations. For example, the #5 key combination enables callers to mute and unmute their phones during a meeting.

PSTN phones use in-band DTMF, which embeds the tone in the audio stream. Although in-band DTMF is efficient, it cannot carry DTMF signals reliably when a voice compression codec is used.

H.323 clients can use out-of-band DTMF, which carries digitized information on a separate data channel and sends this information directly to Cisco Unified MeetingPlace H.323/SIP IP Gateway. Because out-of-band DTMF does not require that the tone be deciphered, distortion and signal loss are minimal.

The Cisco Unified MeetingPlace system also supports RFC 2833: DTMF signals can be sent in the RTP stream by using packets designed to carry the signal characteristics. The DTMF signal is not embedded in the media and, therefore, does not suffer signal loss due to audio compression.

Audio Quality During a Cisco Unified MeetingPlace Meeting

The audio quality during a meeting depends upon the architecture of your network. Severe demands on bandwidth, overloading, and latency cause dropped packets, resulting in broken audio, congestion, and disruption of service.

In general, a switched-100 Mbps network handles VoIP traffic efficiently. To alleviate potentially disruptive service and to improve audio quality, consider implementing class of service (CoS) and quality of service (QoS).

When the server handles over 400 ports of IP calls, voice quality degradation can occur because of network congestion. CoS is a technology that helps manage network traffic by assigning a class to similar types of traffic and assigning a priority to each class. Typically in a VoIP environment, voice traffic is set to a high priority while data traffic is set to a low priority, and CoS makes a best effort to provide QoS by managing traffic based upon the assigned class and priority.

When a call is placed from a PSTN phone to a Cisco IP phone, the call is routed through a voice gateway, which is the demarcation point where the circuit-switched voice network meets the packet-switched data network. The primary responsibility of the voice gateway is to ensure that PSTN voice traffic reaches the data network and vice versa. You can use the voice gateway to forward an IP or PSTN call to its opposing network through Cisco Unified Communications Manager or a PBX.

When a call is placed from an Cisco IP phone, it is routed to Cisco Unified Communications Manager, which is responsible for setting up the call, directing the call to the called device, and sending network information- such as the IP address, UDP port number, and communication capabilities of the called device-to the Cisco IP phone. After receiving the information, the Cisco IP phone sends its digitized voice traffic directly to the called device.

After codec information, IP address, and UDP port number are received, the Cisco IP phone or voice gateway uses the information to send voice traffic directly to the Cisco Unified MeetingPlace Audio Server system. The Cisco IP phone or voice gateway is connected to the Cisco Unified MeetingPlace Audio Server system after each device exchanges data.

After codec information, IP address, and UDP port number of the Cisco Unified MeetingPlace Audio Server system are received, the H.323 device or Cisco SIP IP phone uses the information to send voice traffic directly to the Cisco Unified MeetingPlace Audio Server system. The H.323 device or Cisco SIP IP phone is connected to the Cisco Unified MeetingPlace Audio Server system after each device exchanges data.

Same as for H.323 device.

Notes for Step 2:

The H.323 device or Cisco SIP IP phone and Cisco Unified MeetingPlace H.323/SIP IP Gateway determine if the Cisco Unified MeetingPlace Audio Server system can accept the call. By using the Gateway SIM, the Cisco Unified MeetingPlace H.323/SIP IP Gateway communicates directly with the Cisco Unified MeetingPlace Audio Server system to determine its availability.

If the Cisco Unified MeetingPlace Audio Server system is unavailable, Cisco Unified MeetingPlace H.323/SIP IP Gateway informs the H.323 device or Cisco SIP IP phone, and depending upon system configuration, callers may hear a message informing them that the call cannot be accepted.

Once codec negotiation is complete, Cisco Unified MeetingPlace H.323/SIP IP Gateway retrieves an IP address and UDP port number from the Cisco Unified MeetingPlace Audio Server system by using Gateway SIM. This IP address and UDP port number provide access to the meeting.